Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.

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I am unable to login to Grandstream GXP2170 with username admin and password admin.

How to reset Grandstream GXP2170 to factory defaults?

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Hello Experts,
We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I have strange issue one of my user extension 508 voice mail going into ext 801 voicemail box when using DID but when dial extension 508 it goes into right mailbox of 508, i have checked the user phone settings, DID settings to make sure DID rings and send voicemail to right extension from vicidial gui, here is Dial plan for ext 508 which is setup properly can someone please help me figure this out. Thanks

exten => 3508,1,AGI(agi://
exten => 3508,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/44048020)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100075)
exten => 3508,n,Wait,2
exten => 3508,n,Dial(SIP/508,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100031)
exten => 3508,n,Dial(SIP/508&SIP/801&SIP/505&SIP/507&SIP/509,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/spool/asterisk/voicemail/default/508/unavail)
exten => 3508,n,VoiceMail(508@default)
I am looking for MITEL pbx call accounting software for a time. Through my search i found this website which they have CMar4Pabx call accounting software. Does any one recommend this to me and how good is it ? Your help is greatly appreciated.
We have an old Asterisk (v.2.x) phone server in our office.  I'm new to the system and need to change an extension number from a rapid busy signal to a working extension.  Also, we have several extension that simple hang-up when dialed (no tones of any sort).  How do we edit those extensions?

I'm new to Linux, but I've figured out how to browse directories and edit conf files.
We are in the need of a new phone system and there seems to be a mix of vendors pushing a hosted solution.  Has anyone upgraded in the past couple years to a hosted PBX solution and want to share the experience?  Of course the vendors not offering a hosted solution say to stay away from them they are not reliable.  I understand a lot of the bad from hosted is likely a so so Internet connection.  We have 100x100 fiber so I don't think that would pose an issue.  The big downfall I see is you pay for the hosted forever where an appliance based system, you buy it once and usually good for 10+ years.
Hi, I had a new phone system installed and when I lift the handrest line 2 picks up and not line 1. I need CO1 to pickup by default since thats the line has unlimited calls rather then the 2nd line has pay per call. Can anyone help how to program the phone from picking up line 2 and pick up line 1.
Hi all,

In exchange Unified Messaging if you use Auto Attendant you can't direct extensions directly if you don't press the pound key twice with speech recognition disabled.

I asked Microsoft if it's possible to direct Extension without having to press the # key twice but apparently you can't do so without enabling speech recognition which my customer doesn't want.

Is there any possible way to work around this or do I have to get another IVR system? If so what is the easiest simple to deploy IVR on the market?

Have a client with a Toshiba CIX 100 PBX.  For the most part, have been able to fix some old setup issues with the system.  However. faced with an issue that I can't seem to figure out.  I have experience with VOIP systems and translating back to a digital PBX is causing me some frustration.

When an outside call comes in, all phones ring.  With a VOIP system, I would be looking at a hunt group (or at least the ones I work on).  I need to be able to limit which phones ring on an incoming call. So questions are:

1) By default, does the CIX automatically ring all phones on an incoming call?
2) How do I assign a single phone to an outside line?
3) How do I assign a hunt group to an outside line?
Hello all,

I need to configure the Sonus gateway to route inbound calls to Toll Free number (main number) based on their area code to their particular response group on my Skype for Business server.

For instance, I have a response group for Chicago with DID 312XXXXXX and I have a main toll free number in which case all the calls come to this particular DID (800XXXXXXX) so I need to route all incoming calls from Chicago to the Chicago response group and the same thing for every other state.

I made the area codes already so all calls coming from Chicago state is directly forwarded to the Chicago response groups but the only thing I couldn't do is route all inbound calls coming to the Toll Free number to the response groups.

I would appreciate any advise on this.

I am tasked with setting up a lone staff alert button on a 5320 IP phone that will ring or alert all other phones (12) on the system, is this possible on the mitel 3300 system?
Each phone would hopefully have a key to press that will alert all other phones that the user needs help,

Thank you in advance of any help.

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I'm trying to add a single queue 'csqFabFursEscal' in the working UCCX script (FabFurs_Working.pdf).  I've made changes in an attempt to add the queue in the broken script (FabFurs_broke.pdf) and it doesn't work.  Everytime I load the broken script into my test application I get a system message saying their are problems and to call back.  

This isn't a complex script and I know this is an easy solution for someone better at scripting.  I would really appreciate a UCCX scripting guru take a look.

Has anyone managed to get Cisco 7942G to work with RingCentral.  If so could you share the XML file, ours is just stuck on registering.

On cisco IP phones (model number 7911) it stores some useful information about placed/received calls in the directories application- is this data stored locally on some storage within the phone, or would this be stored in a central database in a managed voip environment, if so being a cisco device can you elaborate where that information may be stored.
Hi all, I am in the process of diverting an incoming call to a particular number on Sonus gateway from a number (Inbound let's say 5000 number) to an external PSTN number 9999 and I would like to do this on the Gateway's end not on the Lync client.

I was able to do this on the client end by simply entering the number in the call forward on client side but if I do this through the gateway with Normalization rule (using Transformation table) with the appropriate Signaling group and Call routing table I get no error in the logs if I choose one ITSP or I get Proxy Authentication Required if I choose another ITSP as signaling group destination.

How do I solve this authentication issue when forwarding calls from a number to another? I have been reading articles but mostly it says it has to do with sip manipulation but I don't want to manipulate the sip, I just want to forward call from a number to another.

I would appreciate any suggestion.
Need 2-line analog phones (not VOIP).  Can't seem to find anything reasonably priced for my client.  Their needs are pretty simple:

1.  Telephones (quantity 4) for two-line analog.
2. Must be able to indicate when a call was missed or message was left, by way of an LED on the phone.
3. Must be able to page the other phones (hands free) by pressing a page button.
4. (optional) Intercom capability.

They currently have 2 of the Cortelco 7-series 2740 phones. These have been replaced by the 2750, and they're over $200 each and take 3 weeks to ship.

Anyone have a better idea for 2-line phones?

We supply a software application that is run on Remote Desktop servers (each client has their own Virtual Server).  One of the areas I am currently researching is to provide the ability for users to click on a "Call" button in our application, and dial the number (whilst connected to the RDP).  I have found a couple of solutions that will allow me to press the "Call" button, which will then dial the recipient's number (along with the user making the call), which in theory, should be ok - but I am yet to test it and am interested in all options.

For me, the perfect solution would be for users to be able to plug in a headset on their local laptop, run the application on the Remote Desktop and call - without impacting server performance in any way.  Or, if they have a phone system already in place - somehow dial through that.

Any pointers, products, suggestions - would be massively appreciated.
Hello all, I am in the process of rolling out a VOIP solution for an office of 15 to 20 lines.  The customer is debating between hosted solution like Zultys Hosted with a 3 year contract for $305 recurring fee and zultys 36G phones for $194 per phone, or go with an appliances  and manage it myself a switchvox E510 for $695. and phones are D60 for $139 each and software registration code for $1000, and extension licence and subscription fee 0f $80 a line per year. I am really leaning towards the appliance but just not sure of the switchvox.  I really need something inexpensive but dependable any suggestions??
Hello We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I am trying to set up my dialplan entry so we can call out from Canada to Irland and Trinidad, our current Dialplan allow us to call only witin north america but not outside,
Here I have the country and area codes that need to be called out from Canada
Irland 353-287-xxx-xxxx
Trinidad 868-788-xxxx

Here I have my Dailplan that i have made up to call out but no luck can someone please guide me correct dialplan please

exten => _9353NXXXXXXXXX,1,AGI(agi://
exten => _91868NXXXXXX,1,AGI(agi://
exten => _9353NXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _91868NXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _9353NXXXXXXXXX,3,Hangup
exten => _91868NXXXXXX,3,Hangup

Here is the call_log but no luck (last 4 digits of phone number being modified to xxxx)

[Nov 1 13:48:07] == Using SIP RTP CoS mark 5
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:1] AGI("SIP/298-00017de5", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Nov 1 13:48:07] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Nov 1 13:48:07] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171101134807_298_1868788xxxx)
[Nov 1 13:48:07] -- <SIP/298-00017de5>AGI Script agi-NVA_recording.agi completed, returning 0
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:2]
Do the Comcast Panasonic KX-TPA65 VOIP phones also have a network jack for plugging in a PC (in addition to plugging the phone into the network)?
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If your answer is factory reset, please skip this question.

User reported a problem for iPhone 7.

On iOS 11.0.2, outgoing phone call  can be made but no incoming phone call could be made.

On iOS 11.0.3, the situation is reverse. No outgoing phone call could be made whereas sometimes incoming phone calls can be made.

However, if the user dials a USSD code which is carrier-specific, it will be OK to dial an outgoing call for once.

Already contacted the carrier and the SIM card is fine on another Android phone or iPhone.

How to fix without factory reset?

Phone call is a basic feature of a smartphone and I have not encountered similar problem in any Android phones.
I need some insights on how to start a call Center for a medical office. The office averages about 350 inbound phone calls per day and 200 outbound calls daily. Any suggestions as of best applications, service providers to use for this project would be greatly appreciated.

When I have someone on hold, and I'm engaging in a consultative transfer (speaking with the person I am about to transfer them to), the headset switches to speakerphone for 1 second every few minutes. It results in me not being able to hear them mid-speech and vice versa; its a real pain. It happens to all 7 of our phones, throughout the building. Have tried speaking with Mitel, system help companies, looking through the set-up manual, looking through the troubleshooting section, googling the exact issue.... etc etc. Nothing! Please can anyone help? My boss has given me the task of "sorting it out" and I just can't seem to :( Thanks, Sophie.
This phone will be exclusively used in Vietnam, 5% Cambodia, 5% Thailand.

Desired specs:
2G or more likely 3G, dual slot, camera, internet capable, touch screen.
Apple and Samsung won't work as they are too expensive.

As far as data for internet I do not know how many GB they would desire (or really need) and I can not pry to much as far as what they would want. The gift is to be a phone far better than the one they have (previous question on Nokia 150).
CDMA is not used outside of North America, so you will not get any coverage at all. Your CDMA phone will useless in Vietnam or prettywell anywhere outside of the USA and Canada (and even there CDMA is a dying standard). You will need a GSM phone and one that operates on non-North American frequencies; i.e. a quad-band phone.
I'll later post the specs of my Kata phone since they told me that my phone was quite better than theirs.
Here they are:
Kata V4 - 4.5-inch IPS Quad Core International Unlocked Smartphone Android 5.1 - Dark Grey - Super Slim HD 1.3 GHz Dual Sim Card GSM 8MP Camera
GSM Quadband 850/900/1800/1900, 3G WCMA 850/1900/2100, Android 5.1 (Lollipop)
4.5" IPS FWVGA Display: (480 x 854) with 1.3 GHz Quad-Core Processor
8MP Camera with LED Flash with HD recording , 5MP Front Camera for Video Calling and instant self portraits
8GB ROM, 1GB RAM with 32GB expandable storage …
I thought I had found an optimal solution in Possio Greta:

But I talked to the head office in Sweden, and it turned out that it doesn't work everywhere in the world. I can't remember the details now, but I had a lengthy discussion about it, and it's of quite limited use for me when I travel between different parts of the world and different countries.

Otherwise, is there any alternative to smartphone and internet, such as a small portable fax machine or any other device or other type of communication (analogue instead of digital etc.)? Telex, satellite, short-wave radio, SIP for fax etc.?
I have a pair of C-level users who both experienced a problem at one of my sites. I'm trying to determine if it's a GoToMeeting problem (not my problem) or a phone problem (definitely my problem). You guys will give me a quick answer, I just know it. :-)

When the users join the audio portion of a GoToMeeting event from their Cisco desk phones (on my CUCM-powered phone system), right after the point where they enter their PIN number for the meeting, they are supposed to press the pound key "#" to join the audio part of the meeting. Every time each of them presses the "#" key on their desk phones, either the meeting or the phone hangs up the call. When they join using their cell phones it works fine.

This just started happening and no changes have recently been made to the phones or the CUCM settings. I'm told that it happened once before but was not reported, and that other occasion occurred several months ago but then the conditions returned to normal operation and they didn't see this again until yesterday.

Can I get some opinions on which party is responsible?

Thanks experts!


Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.